use anyhow::Result;
use bytes::BytesMut;
use clap::Parser;
use env_logger::Target;
use log::{debug, error, trace};
use rtc::interceptor::Registry;
use rtc::peer_connection::RTCPeerConnectionBuilder;
use rtc::peer_connection::configuration::RTCConfigurationBuilder;
use rtc::peer_connection::configuration::interceptor_registry::register_default_interceptors;
use rtc::peer_connection::configuration::media_engine::{
MIME_TYPE_OPUS, MIME_TYPE_VP8, MediaEngine,
};
use rtc::peer_connection::configuration::setting_engine::SettingEngine;
use rtc::peer_connection::event::RTCTrackEvent;
use rtc::peer_connection::event::{RTCEvent, RTCPeerConnectionEvent};
use rtc::peer_connection::sdp::RTCSessionDescription;
use rtc::peer_connection::state::RTCPeerConnectionState;
use rtc::peer_connection::transport::RTCDtlsRole;
use rtc::peer_connection::transport::RTCIceServer;
use rtc::peer_connection::transport::{CandidateConfig, CandidateHostConfig, RTCIceCandidate};
use rtc::rtcp::payload_feedbacks::picture_loss_indication::PictureLossIndication;
use rtc::rtp_transceiver::rtp_sender::RTCRtpCodecParameters;
use rtc::rtp_transceiver::rtp_sender::RtpCodecKind;
use rtc::sansio::Protocol;
use rtc::shared::error::Error;
use rtc::shared::marshal::Marshal;
use rtc::shared::{TaggedBytesMut, TransportContext, TransportProtocol};
use signal;
use std::collections::HashMap;
use std::fs::OpenOptions;
use std::io::Write;
use std::str::FromStr;
use std::time::{Duration, Instant};
use tokio::net::UdpSocket;
use tokio::sync::mpsc::channel;
const DEFAULT_TIMEOUT_DURATION: Duration = Duration::from_secs(86400);
#[derive(Parser)]
#[command(name = "rtp-forwarder")]
#[command(author = "Rusty Rain <y@liu.mx>")]
#[command(version = "0.1.0")]
#[command(about = "An example of RTP forwarder")]
struct Cli {
#[arg(short, long)]
debug: bool,
#[arg(short, long, default_value_t = format!("INFO"))]
log_level: String,
#[arg(short, long, default_value_t = format!(""))]
input_sdp_file: String,
#[arg(short, long, default_value_t = format!(""))]
output_log_file: String,
}
#[tokio::main]
async fn main() -> Result<()> {
let cli = Cli::parse();
let input_sdp_file = cli.input_sdp_file;
let output_log_file = cli.output_log_file;
let log_level = log::LevelFilter::from_str(&cli.log_level)?;
if cli.debug {
env_logger::Builder::new()
.target(if !output_log_file.is_empty() {
Target::Pipe(Box::new(
OpenOptions::new()
.create(true)
.write(true)
.truncate(true)
.open(output_log_file)?,
))
} else {
Target::Stdout
})
.format(|buf, record| {
writeln!(
buf,
"{}:{} [{}] {} - {}",
record.file().unwrap_or("unknown"),
record.line().unwrap_or(0),
record.level(),
chrono::Local::now().format("%H:%M:%S.%6f"),
record.args()
)
})
.filter(None, log_level)
.init();
}
println!("Paste your offer here:");
let line = if input_sdp_file.is_empty() {
signal::must_read_stdin()?
} else {
std::fs::read_to_string(&input_sdp_file)?
};
let desc_data = signal::decode(line.as_str())?;
let offer = serde_json::from_str::<RTCSessionDescription>(&desc_data)?;
println!("Offer received: {}", offer);
let audio_socket = UdpSocket::bind("127.0.0.1:0").await?;
audio_socket.connect("127.0.0.1:4000").await?;
println!("Audio will be forwarded to 127.0.0.1:4000");
let video_socket = UdpSocket::bind("127.0.0.1:0").await?;
video_socket.connect("127.0.0.1:4002").await?;
println!("Video will be forwarded to 127.0.0.1:4002");
run_peer_connection(offer, audio_socket, video_socket).await?;
Ok(())
}
async fn run_peer_connection(
offer: RTCSessionDescription,
audio_socket: UdpSocket,
video_socket: UdpSocket,
) -> Result<()> {
let socket = UdpSocket::bind("127.0.0.1:0").await?;
let local_addr = socket.local_addr()?;
let mut setting_engine = SettingEngine::default();
setting_engine.set_answering_dtls_role(RTCDtlsRole::Server)?;
let mut media_engine = MediaEngine::default();
media_engine.register_codec(
RTCRtpCodecParameters {
rtp_codec: rtc::rtp_transceiver::rtp_sender::RTCRtpCodec {
mime_type: MIME_TYPE_VP8.to_string(),
clock_rate: 90000,
channels: 0,
sdp_fmtp_line: "".to_string(),
rtcp_feedback: vec![],
},
payload_type: 96,
..Default::default()
},
RtpCodecKind::Video,
)?;
media_engine.register_codec(
RTCRtpCodecParameters {
rtp_codec: rtc::rtp_transceiver::rtp_sender::RTCRtpCodec {
mime_type: MIME_TYPE_OPUS.to_string(),
clock_rate: 48000,
channels: 2,
sdp_fmtp_line: "".to_string(),
rtcp_feedback: vec![],
},
payload_type: 111,
..Default::default()
},
RtpCodecKind::Audio,
)?;
let registry = Registry::new();
let registry = register_default_interceptors(registry, &mut media_engine)?;
let config = RTCConfigurationBuilder::new()
.with_ice_servers(vec![RTCIceServer {
urls: vec!["stun:stun.l.google.com:19302".to_string()],
..Default::default()
}])
.build();
let mut peer_connection = RTCPeerConnectionBuilder::new()
.with_configuration(config)
.with_setting_engine(setting_engine)
.with_media_engine(media_engine)
.with_interceptor_registry(registry)
.build()?;
peer_connection.add_transceiver_from_kind(RtpCodecKind::Audio, None)?;
peer_connection.add_transceiver_from_kind(RtpCodecKind::Video, None)?;
peer_connection.set_remote_description(offer)?;
let candidate = CandidateHostConfig {
base_config: CandidateConfig {
network: "udp".to_owned(),
address: local_addr.ip().to_string(),
port: local_addr.port(),
component: 1,
..Default::default()
},
..Default::default()
}
.new_candidate_host()?;
let local_candidate_init = RTCIceCandidate::from(&candidate).to_json()?;
peer_connection.add_local_candidate(local_candidate_init)?;
let answer = peer_connection.create_answer(None)?;
peer_connection.set_local_description(answer.clone())?;
println!("RTP forwarder listening on {}...", socket.local_addr()?);
let json_str = serde_json::to_string(&answer)?;
let b64 = signal::encode(&json_str);
println!("\nPaste this answer in your browser:\n{}\n", b64);
let (_event_tx, mut event_rx) = channel::<RTCEvent>(8);
let mut buf = vec![0; 2000];
let mut pli_last_sent = Instant::now();
let mut ssrc2kind: HashMap<u32, RtpCodecKind> = HashMap::new(); let audio_payload_type = 111u8;
let video_payload_type = 96u8;
println!("Press Ctrl-C to stop");
'EventLoop: loop {
while let Some(msg) = peer_connection.poll_write() {
match socket.send_to(&msg.message, msg.transport.peer_addr).await {
Ok(n) => {
trace!(
"socket write to {} with {} bytes",
msg.transport.peer_addr, n
);
}
Err(err) => {
error!("socket write error: {}", err);
}
}
}
while let Some(event) = peer_connection.poll_event() {
match event {
RTCPeerConnectionEvent::OnConnectionStateChangeEvent(state) => {
println!("Peer Connection State: {}", state);
if state == RTCPeerConnectionState::Failed {
println!("Connection failed, exiting...");
break 'EventLoop;
} else if state == RTCPeerConnectionState::Connected {
println!("Connection established!");
}
}
RTCPeerConnectionEvent::OnTrack(RTCTrackEvent::OnOpen(init)) => {
println!(
"OnTrack::OnOpen - receiver_id: {:?}, track_id: {}",
init.receiver_id, init.track_id
);
if let Some(receiver) = peer_connection.rtp_receiver(init.receiver_id) {
let track = receiver.track();
println!(
"Track kind: {}, codec: {}",
track.kind(),
track
.codec(
track
.ssrcs()
.next()
.ok_or(Error::ErrRTPReceiverForSSRCTrackStreamNotFound)?,
)
.ok_or(Error::ErrCodecNotFound)?
.mime_type
);
ssrc2kind.insert(
track
.ssrcs()
.last()
.ok_or(Error::ErrRTPReceiverForSSRCTrackStreamNotFound)?,
track.kind(),
);
}
}
RTCPeerConnectionEvent::OnTrack(RTCTrackEvent::OnClose(_track_id)) => {}
_ => {}
}
}
while let Some(message) = peer_connection.poll_read() {
match message {
rtc::peer_connection::message::RTCMessage::RtpPacket(_track_id, mut rtp_packet) => {
let kind = ssrc2kind
.get(&rtp_packet.header.ssrc)
.ok_or(Error::ErrTrackNotExisted)?;
let target_socket = if kind == &RtpCodecKind::Video {
rtp_packet.header.payload_type = video_payload_type;
&video_socket
} else {
rtp_packet.header.payload_type = audio_payload_type;
&audio_socket
};
let mut marshal_buf = vec![0u8; 1500];
if let Ok(n) = rtp_packet.marshal_to(&mut marshal_buf) {
if let Err(err) = target_socket.send(&marshal_buf[..n]).await {
if !err.to_string().contains("Connection refused") {
error!("Forward {} error: {}", kind, err);
}
} else {
trace!("Forwarded {} packet, {} bytes", kind, n);
}
}
}
rtc::peer_connection::message::RTCMessage::RtcpPacket(_, _) => {
trace!("Received RTCP packets");
}
rtc::peer_connection::message::RTCMessage::DataChannelMessage(_, _) => {}
}
}
let eto = peer_connection
.poll_timeout()
.unwrap_or(Instant::now() + DEFAULT_TIMEOUT_DURATION);
let delay_from_now = eto
.checked_duration_since(Instant::now())
.unwrap_or(Duration::from_secs(0));
if delay_from_now.is_zero() {
peer_connection.handle_timeout(Instant::now())?;
continue;
}
let timer = tokio::time::sleep(delay_from_now);
tokio::pin!(timer);
tokio::select! {
biased;
_ = tokio::signal::ctrl_c() => {
println!("\nCtrl-C received, shutting down...");
break 'EventLoop;
}
res = event_rx.recv() => {
match res {
Some(event) => {
peer_connection.handle_event(event)?;
}
None => {
eprintln!("event_rx closed");
break 'EventLoop;
}
}
}
_ = timer.as_mut() => {
let now = Instant::now();
peer_connection.handle_timeout(now)?;
if now > pli_last_sent + Duration::from_secs(3) {
for (ssrc, kind) in &ssrc2kind {
debug!("Sending PLI for {} track (SSRC: {})", kind, ssrc);
let receiver_ids: Vec<_> = peer_connection.get_receivers().collect();
for receiver_id in receiver_ids {
if let Some(mut rtp_receiver) = peer_connection.rtp_receiver(receiver_id) {
let _ = rtp_receiver.write_rtcp(vec![Box::new(PictureLossIndication {
sender_ssrc: 0,
media_ssrc: *ssrc,
})]);
}
}
}
pli_last_sent = now;
}
}
res = socket.recv_from(&mut buf) => {
match res {
Ok((n, peer_addr)) => {
trace!("socket read {} bytes from {}", n, peer_addr);
peer_connection.handle_read(TaggedBytesMut {
now: Instant::now(),
transport: TransportContext {
local_addr,
peer_addr,
ecn: None,
transport_protocol: TransportProtocol::UDP,
},
message: BytesMut::from(&buf[..n]),
})?;
}
Err(err) => {
eprintln!("socket read error {}", err);
break 'EventLoop;
}
}
}
}
}
peer_connection.close()?;
println!("Event loop exited");
Ok(())
}